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* Don't use the mutex in the base getClockLatency implementationChris Robinson2017-02-281-3/+8
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* Print WARNs when a device or context error is generatedChris Robinson2017-02-271-0/+1
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* Use separate enums for the ambisonic channel order and normalizationChris Robinson2017-02-273-11/+24
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* Move the current buffer queue entry and play position to the voiceChris Robinson2017-02-273-25/+24
| | | | | | | | | | | | | | This has a couple behavioral changes. First and biggest is that querying AL_BUFFERS_PROCESSED from a source will always return all buffers processed when in an AL_STOPPED state. Previously all buffers would be set as processed when first becoming stopped, but newly queued buffers would *not* be indicated as processed. That old behavior was not compliant with the spec, which unequivocally states "On a source in the AL_STOPPED state, all buffers are processed." Secondly, querying AL_BUFFER on an AL_STREAMING source will now always return 0. Previously it would return the current "active" buffer in the queue, but there's no basis for that in the spec.
* Ensure a non-playing or -paused source does not use a mixing voiceChris Robinson2017-02-252-2/+6
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* Remove an unused functionChris Robinson2017-02-232-6/+0
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* Remove CalcXYZCoeffs and inline CalcAngleCoeffsChris Robinson2017-02-235-21/+13
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* Alter how panpot/pair-wise panning worksChris Robinson2017-02-232-32/+22
| | | | | | | | | | | | | | | | | | | This change allows pair-wise panning to mostly go through the normal ambisonic panning methods, with one special-case. First, a term is added to the stereo decoder matrix's X coefficient so that a centered sound is reduced by -3dB on each output channel. Panning in front creates a similar gain response to the typical L = sqrt(1-pan) R = sqrt(pan) for pan = [0,1]. Panning behind the listener can reduce (up to) an additional -10dB, creating a audible difference between front and back sounds as if simulating head obstruction. Secondly, as a special-case, the source positions are warped when calculating the ambisonic coefficients so that full left panning is reached at -30 degrees and full right at +30 degrees. This is to retain the expected 60-degree stereo width. This warping does not apply to B-Format buffer input, although it otherwise has the same gain responses.
* Rename stereo-panning option to stereo-encodingChris Robinson2017-02-222-11/+11
| | | | Also rename the 'paired' value to 'panpot', and make it the default.
* Limit filter gains to -24dBChris Robinson2017-02-224-39/+33
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* Reduce the default period count to 3Chris Robinson2017-02-221-1/+1
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* Don't remove a period from the OSS bufferChris Robinson2017-02-221-11/+4
| | | | | Since we're now waiting for space to be available before mixing, the mixing buffer isn't adding another period.
* Fix OpenSL latency calculationChris Robinson2017-02-221-2/+2
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* Reduce some codeChris Robinson2017-02-211-61/+33
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* Make the "sends" config option act as a limitChris Robinson2017-02-211-8/+16
| | | | | Instead of forcing the device to always use the specified send count, it simply limits requests to it.
* Increase the default effect slot and send countChris Robinson2017-02-211-13/+15
| | | | | | | | | | The default number of auxiliary effect slots is now 64. This can still be raised by the config file without a hard maximum, but incurs processing cost for each effect slot generated by the app. The default number of source sends is now actually 2, as per the EFX docs. However, it can be raised up to 16 via ALC_MAX_AUXILIARY_SENDS attribute requests, rather than the previous 4.
* Dynamically allocate the ALsource Send[] arrayChris Robinson2017-02-212-45/+59
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* Interleave the voice and source property objectsChris Robinson2017-02-211-13/+12
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* Make the voices' Send[] array dynamically sizedChris Robinson2017-02-212-23/+39
| | | | | The voices are still all allocated in one chunk to avoid memory fragmentation. But they're accessed as an array of pointers since the size isn't static.
* Print warnings about missing libraries and functionsChris Robinson2017-02-213-2/+24
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* Avoid duplicating device buffer layout logicChris Robinson2017-02-203-38/+38
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* Remove an unused flag enumChris Robinson2017-02-201-3/+0
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* Allow distance compensation for non-HQ rendering as wellChris Robinson2017-02-201-37/+44
| | | | | It still requires a custom configuration to specify appropriate speaker distances.
* Remove the separate surround51rear decoder optionChris Robinson2017-02-191-2/+2
| | | | | | Both 5.1 Side and Rear configurations use 'surround51' to look up the appropriate decoder file. The decoder loader already handles mapping between rear and side channels, so there's no need for separate options.
* Apply distance compensation when writing to the outputChris Robinson2017-02-194-122/+94
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* Don't use periphonic FOA when the HOA decoder is not periphonicChris Robinson2017-02-195-41/+55
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* Remove the sinc8 resampler optionChris Robinson2017-02-196-286/+26
| | | | | Perf shows less than 1 percent CPU difference from the higher quality bsinc resampler, but uses almost twice as much memory (a 128KB lookup table).
* Always lock the device backend before calling aluMixDataChris Robinson2017-02-1811-50/+63
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* Return some device latency by defaultChris Robinson2017-02-181-2/+7
| | | | | | | A device will never have 0 latency. OpenAL Soft itself uses a sample buffer length of UpdateSize*NumUpdates, and during playback will have about (NumUpdates-1) periods filled, more or less. Without a more accurate measurement from the playback system, this is better than reporting 0.
* Use select() to wait for audio with OSS and SolarisChris Robinson2017-02-182-85/+137
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* Reorganize ALvoice membersChris Robinson2017-02-152-88/+95
| | | | | This places the Send[] array at the end of the struct, making it easier to handle dynamically.
* Make ALsourceProps' Send array dynamically sizedChris Robinson2017-02-143-20/+102
| | | | | | ALsourceProps' Send[] array is placed at the end of the struct, and given an indeterminate size. Extra space is allocated at the end of each struct given the number of auxiliary sends set for the device.
* Make the source state atomicChris Robinson2017-02-133-7/+8
| | | | | Since it's modified by the mixer when playback is ended, a plain struct member isn't safe.
* Put BsincState in a generic unionChris Robinson2017-02-139-75/+75
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* Porperly check for and use __builtin_assume_alignedChris Robinson2017-02-132-12/+20
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* Clean up the bsinc mixer a bitChris Robinson2017-02-123-26/+28
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* Add NEON-enhanced resamplersChris Robinson2017-02-123-4/+294
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* Fix more uses of unsigned sizes and offsetsChris Robinson2017-02-102-9/+9
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* Remove a couple context lock wrapper functionsChris Robinson2017-02-071-3/+0
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* Properly capitalize NEONChris Robinson2017-02-071-1/+1
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* Clear trailing whitespace from the cpu features stringChris Robinson2017-02-071-0/+5
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* Use the correct IID for the opensl buffer queueChris Robinson2017-02-071-2/+3
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* Convert the OpenSL backend to the new backend APIChris Robinson2017-02-073-177/+434
| | | | | This also removes the buffer queue callback's call to aluMixData, which could potentially block on a mutex.
* Fix for NULL JNIEnvChris Robinson2017-02-052-5/+9
| | | | Which can happen with native-only apps
* Call getSystemService as a non-static functionChris Robinson2017-02-031-4/+4
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* Android's AudioManager.getProperty(String) returns a StringChris Robinson2017-02-021-1/+1
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* Replace more ALuint with ALsizeiChris Robinson2017-01-293-15/+15
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* Fix late reverb low-pass filteringChris Robinson2017-01-291-1/+1
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* Use an all-pass series on each late reverb lineChris Robinson2017-01-281-261/+396
| | | | | | | | | This attempts to improve the smoothness of the late reverb decay by passing each line through multiple all-pass filters. Some work is still needed to work better in high-density and not-so-high-diffusion environments. This also removes the decay from the early reflections, since it's no longer continuous feedback.
* Add more traces for the Java calls being madeChris Robinson2017-01-271-7/+22
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